Step size convergence control

ABSTRACT

A method of converging a step size control for an adaptive filter of a communication channel is disclosed. This method has the steps of: (1) initializing a nominal step size value and a penalty point value; (2) combining the nominal step size value and the penalty point value to generate a step size value; and (3) dynamically changing the step size value in response to a characteristic measure of a quality of the communication channel. With this method, the step size value is changed by adjusting the nominal step size value, the penalty point value, or both the nominal step size value and the penalty point value. In a preferred embodiment of the invention, the step size value is decreased by adjusting the penalty point value when: (1) a tone originating from the far end of the communication channel is detected, to prevent the adaptive filter from diverging; (2) full convergence is achieved; (3) a power level of a residual error signal, P e , is less than −60 dBm0 or of a far-end channel signal, P x , is less than −45 dBm0; (4) a level of the channel&#39;s near-end background noise is high; and (5) weak double-talk in the communication channel is detected. The step size value is decreased by adjusting the nominal step size value when an achieved initial combined loss is about 15 dB or greater. On the other hand, the step size value is reset upon an adaptive filter reset that may be triggered by divergence. Additionally, the step size value is reinitialized at the beginning of every forty sample block period, which is 5 ms when an 8 kHz sampling rate is used.

CROSS REFERENCE TO RELATED APPLICATIONS

[0001] Not applicable.

FIELD OF THE INVENTION

[0002] The present invention relates to the reduction of echo signals ina telecommunication link. This is accomplished by better adaptivelymatching the echo canceller characteristics to the transmission pathcharacteristics. More specifically, a method of controlling a step sizefor an adaptive filter of an echo canceller is taught.

BACKGROUND OF THE INVENTION

[0003] Referring to FIG. 1, a representative implementation of atelephone network link 1 is illustrated. Hybrid circuit 2 connects anear-end user telephone 5 to the network 1 and hybrid circuit 9 connectsa far-end user 11 to the network 1. Since the trunk 13 interconnectingthe near-end central office 12 and the far-end central office 14 conveysdigital communications, analog-to-digital (A/D) converters 3 and 10connect the transmitter side of the hybrid circuits at each end of thenetwork, to the digital trunk circuit 13. Similarly, digital-to-analog(D/A) converters 4 and 8 connect the receiver side of the hybrid circuitto the digital trunk circuit 13. Using this network structure, anend-to-end voice communication may take place between two end-userdevices 5 and 11 of the telephone network.

[0004] Each hybrid circuit 2, 9 is a converter that interconnects atwo-wire circuit of the telephone to a four-wire circuit of the centraloffice 12, 14. Both the two-wire and four-wire circuits support thesimultaneous communication of transmit and receive signals. However, thefour-wire circuit of the Public Switched Telephone Network (PSTN) usesone wire pair for the transmit signal and the other wire pair for thereceive signal, while the two-wire circuit must carry both the transmitand receive signals on a single wire pair. Because the transmit andreceive signals are duplexed on the single wire pair, part of thetransmitted signal energy 18 and/or 16 originating from the telephone 5and/or 11 can be reflected back to the telephone by the hybrid circuits9 and 2, respectively. This reflected energy, though delayed in time,substantially replicates the transmitted signal and causes undesirableinterference. When the transmitted signal is human speech, the speakermay hear his or her own speech in the receiver as a delayed andattenuated echo.

[0005] For example, when a user speaks into telephone 5 the voice signalenergy is transmitted to telephone 11 through hybrid 9. Echo is createdwhen the transmitted signal is reflected back by the hybrid circuit 9and passes through the PSTN to the originating telephone. This echo isannoying to the users of the communication link.

[0006] The quality of the communication link may be improved bysubtracting a replica of the originally transmitted signal from the echosignal generated by the hybrid circuit at the far-end of thecommunication link. Since the signal transmitted by the hybrid circuitat the far end contains both the echo and the far-end user's voicesignal of interest, subtracting the replica of the originallytransmitted near-end signal from the transmitted far-end signal willreduce or eliminate the echo, and retain the far-end user's voicesignal.

[0007] As illustrated in FIG. 1, signal splitter 15 provides the signal18 a to D/A 8 and to filter 19 which provides a negative replica −18 ofthe originally transmitted signal 18 to a combiner circuit 17 at thesame time the combined echo signal 18′ and far-end signal of interest 16is provided to the combiner circuit 17. The echo canceller circuit 7contains amplification/attenuation circuitry that attempts to match theamplitude of the replica signal −18 with that of the echo signal 18′. Byprecisely matching the absolute values of the signal amplitudes of thenegative replica −18 and the echo signal 18′, as they are provided tothe combiner circuit 17, the echo may be removed entirely from thesignal 16 a received by the near-end user. However, the complete removalof the echo only occurs under ideal conditions. After being summed bythe combiner 17, the sum of the signal energies is conveyed to thenear-end telephone receiver as signal 16 a.

[0008] A real-world implementation of the communication link representedby FIG. 1 does not provide the ideal conditions needed to entirelyeliminate the signal. The original signal information contained in theecho signal 18′, which is received by the combiner 17, is distorted bythe non-linearities present in the A/D and D/A conversions that theoriginal signal 18 has undergone.

[0009] The μ-law or A-law A/D and D/A conversions experienced within thetransmission path are nonlinear in nature and present a significantproblem to the linear adaptive filter 19 typically used in echocancellers. Additional signal distortion is caused by the non-linearityof the hybrid circuit 9. The linear adaptive filter 19 cannot match thenon-linear distortions introduced by the μ-law or A-law conversions andthe hybrid circuit 9 and, as a result, cannot cancel them. Therefore, atypical voice communication link over the PSTN is subject to echo thatcannot be completely cancelled using conventional approaches, such aslinear adaptive filtering. In essence, the echo canceller synthesizesthe estimated echo, which is subtracted from the composite signal (16,18′) of the combined far-end signal of interest 16 and echo 18′.Together, the signal distortion caused by the non-linearities of themultiple conversions and the inability of the echo canceller toprecisely model the true echo path limit the realizable echo rejection.

SUMMARY OF THE INVENTION

[0010] The present invention provides a substantial improvement over theprior art in the reduction of an echo signal in a telecommunicationlink. This is accomplished by better adaptively matching the echocanceller characteristics to the transmission path characteristics.

[0011] The present invention discloses a method of converging a stepsize control for an adaptive filter of a communication channel. Thismethod has the steps of: (1) initializing a nominal step size value anda penalty point value; (2) combining the nominal step size value and thepenalty point value to generate a step size value; and (3) dynamicallychanging the step size value in response to a characteristic measure ofa quality of the communication channel. With this method, the step sizevalue is changed by adjusting the nominal step size value, the penaltypoint value, or both the nominal step size value and the penalty pointvalue. In a preferred embodiment of the invention, the step size valueis decreased by adjusting the penalty point value when: (1) a toneoriginating from the far end of the communication channel is detected,to prevent the adaptive filter from diverging; or (2) full convergenceis achieved; or (3) a power level of a residual error signal, P_(e), isless than −60 dBm0 or (4) a power level of a far-end channel signal,P_(x), is less than −45 dBm0; or (5) a level of the channel's near-endbackground noise is high; or (6) weak double-talk in the communicationchannel is detected. The step size value is decreased by adjusting thenominal step size value when an achieved initial combined loss is about15 dB or greater. On the other hand, the step size value is reset upondetection of an adaptive filter reset that may be triggered bydivergence between the adaptive filter and a speech signal in thecommunication channel. Additionally, the step size value isreinitialized at the beginning of every forty sample block period, whichis 5 ms when an 8 kHz sampling rate is used.

BRIEF DESCRIPTION OF THE DRAWINGS

[0012] Preferred embodiments of the invention are discussed hereinafterin reference to the drawings, in which:

[0013]FIG. 1—illustrates a representative implementation of a telephonenetwork link; and

[0014]FIG. 2—illustrates a representative echo canceller circuitimplemented by an adaptive least mean square algorithm device having acombiner and a digital transversal filter.

DETAILED DESCRIPTION OF PREFERRED EMBODIMENTS

[0015]FIG. 2 illustrates a representative echo canceller circuit 20implemented by an adaptive least mean square (LMS) algorithm devicehaving a combiner 21 and a digital transversal filter 22. The transfercharacteristic of the echo canceller circuit 20 may be expressed by theequation:

e[m]=y[m]−x[m]* h _(k) [m]

[0016] where * is the convolution operator and h_(k)[m] are the filtercoefficients for the k^(th) iteration of the LMS algorithm. The filtercoefficients are generated by the equation:

h_(k+1) [n]=h _(k) [n]+μ·e[m]·x[m−n]

[0017] where e[m] is the error signal 23, x[m] is the far-end excitation24, and μ is the step size. The step size for the normalized LMSalgorithm can be expressed as:$\mu = \frac{2 \cdot \mu_{0}}{{f\left( P_{x} \right)} \cdot {g(N)}}$

[0018] where μ₀ is the nominal step size, P_(x) is the far-end power, Nis the filter length in taps, andf( ) and g( ) are suitable monotone,non-decreasing functions.

[0019] The step size regulates the convergence of the adaptive filtercharacteristics and the transmission path characteristics in followingways. A small step size value provides greater mathematical stability inthe algorithm but slows the rate of convergence. A large step size valueprovides faster convergence but less stability. For arbitrary speechsignals, it is not possible to determine a single value of the nominalstep size that would always work best. The speech power is not constantand may change with every syllable. Similarly, the background noiseduring the conversation may vary as well. Prior art fixed-pointalgorithms have difficulty due to quantization effects in converging thefilter and transmission path characteristics whenever the signal levelsbecome very low and also whenever substantial near-end noise is present.Additionally, when people on both ends of the communication link talk atthe same time (double-talk) the convergence algorithm may need to betemporarily discontinued or slowed.

[0020] For these reasons, the present invention adjusts the step size inreal-time based on several conditions. Control of the step sizeadjustment is bifurcated into a nominal step size selection and a stepsize correction.

[0021] The nominal step size selection is implemented through a finitestate machine. A large nominal step size is used during the initialconvergence. After a certain level of convergence is achieved, thenominal step size is reduced. Stated in other words, an aggressive stepsize is used until the adaptive algorithm makes significant progress inconverging the filter and transmission path characteristics. Thereafter,a more conservative value of the nominal step size is used.

[0022] The two states of the state machine are termed the small andlarge step size states. A conservative value of the nominal step size,for example μ_(0,s)=0.125, is used when the state machine operates inthe small state and a more aggressive value, such as μ_(0,l)=0.25, isused when in the large state of operation. The initial state ofoperation for the finite state machine is the large state. When thecombined loss exceeds 15 dB, the state changes to the small state. Atransition from the small to the large state can also be implemented forvarious situations, such as a filter and channel characteristicdivergence, a filter-reset operation, an echo path change detected, etc.The nominal step size selection may be generalized to use additionalstates if more precise control of the rate of convergence andalgorithmic stability is needed.

[0023] Since there are many other independent conditions that mayrequire changes in step size, the present invention also adjusts thestep size through a system of “penalty points.” Applying a base twologarithm to the previous equation for the step size, μ, and assigningthe result to a variable, m, provides the following equation:

m=log ₂(μ)=1+m ₀ −p _(x) −n

[0024] where m₀=log₂(μ₀), p_(x)=log₂(f(P_(x))), and n=log₂(g(N)). Forthe purpose of describing the invention with simplicity, assume:

f(x)=2^(└log) ^(₂) ^((x)┘) and g(x)=2^(┐log) ^(₂) ^((x)┌).

[0025] The penalty points, ρ, will be subtracted from the step size, asexpressed by the equation:

m=log ₂(μ)=1=m ₀ −p _(x) −n−ρ

[0026] Penalty points may have positive or negative integer values. Apositive penalty point has the effect of decreasing the step size, anegative penalty point has the effect of increasing the step size, and azero penalty point has no effect on the step size.

[0027] Penalty points are used to adjust the step size whenever ashort-term change is needed. The nominal step size is fundamental to theconvergence algorithm and changes to this value should only be madewhen: (1) the filter length, N, changes; (2) the far-end power, P_(x)changes; or (3) initial convergence is achieved. Temporary changes tothe step size should be segregated from the algorithm's fundamentalproperties to allow orthogonality of the algorithm. In other words,segregating the short- and long-term adjustments to the step sizeprovides a modeling, design, and implementation capability that isindependent of the nominal step size control.

[0028] In a preferred embodiment of the invention, with reference toFIG. 2, the penalty points are adjusted under the following conditions.A positive penalty point is added when a tone is detected in the far-endexcitation 24. Decreasing the step size in this instance helps tominimize the extent to which the adaptive filter might diverge. Apositive penalty point is added when full convergence is achieved. Whenthe adaptive filter has fully converged on the voice signal, decreasingthe step size increases the stability of the achieved convergence. Ahigh achieved combined loss is indicative of full convergence.International Telecommunication Union (ITU-T) Recommendation G. 168,which is hereby incorporated by reference into this application,provides another measure of full convergence. Full convergence may alsobe described as the point where further convergence is constrained bythe hardware limitations or where further convergence would be audiblyimperceptible to the users of the telephone link. Other ways of definingconvergence that are obvious to one of ordinary skill in the art, basedupon this disclosure, may be used and are deemed to be part of thisdisclosure.

[0029] In a preferred illustrated embodiment of the invention, apositive penalty point is added when the residual p_(e) or far-endexcitation p_(x) levels are very low. For example, when the echocanceller's residual error power, p_(e), is less than −60 dBm0 or thefar-end excitation power, p_(x), is less than −45 dBm0 the filter isbetter able to converge and the decreased step size will betterstabilize the convergence. One or more penalty points are added when thebackground noise originating from the near-end terminal 25 is moderateor high. For example, two points can be added for high noise and onepoint can be added for moderate background noise. In an exemplaryembodiment, moderate noise has a level of between about −55 dBm0 and −45dBm0 and high noise has a level above about −45 dBm0 or more. Similarly,one or more penalty points are added when weak double-talk is detectedin the link. This condition occurs when the amplitude of the near-endtalker's voice has a much lower level than the far-end talker's voiceand may be detected by the presence of a near-end speech level that isconsiderably above the noise floor when far-end speech is present. Forexample, weak double-talk may also be described as a condition where thenear-end speech signal is about 6 dB below the far-end speech signal,but still 12 dB above the noise floor.

[0030] Other values of penalty points may be added or subtracted in theexamples provided above, since the values indicated are exemplary andnot limiting. Additionally, negative penalty points may be added toincrease the convergence rate. The conditions for adding negativepenalty points may be the inverse of those discussed above for addingpositive points or may be some other identifiable condition. Forexample, negative penalty points are appropriate in a situation wherethe nominal step size is adjusted for a very conservative environmentthat assumes the constant presence of high noise levels.

[0031] In a preferred embodiment, the total number of penalty points islimited to a maximum value to prevent the rate of convergence fromfalling below a particular level. Moreover, the number of penalty pointsis reinitialized to zero periodically. Preferably, the value of thepenalty points, ρ, is recalculated for each of the forty samples takenin a sampling block, which is a 5 ms period using a 8 kHz sampling rate.For every block, the penalty points are initialized to zero. However,the block period, sampling rate, reinitialization period, andrecalculation period may have other values to suit the particularapplication of the invention. The important aspect of dynamicallyadjusting the penalty points is to adjust them based on the changes inthe environment of the communication link and the achieved performanceof the adaptive filter.

[0032] Because many varying and different embodiments may be made withinthe scope of the inventive concept herein taught, and because manymodifications may be made in the embodiments herein detailed inaccordance with the descriptive requirements of the law, it is to beunderstood that the details herein are to be interpreted as illustrativeand not in a limiting sense.

What is claimed is:
 1. A method of converging an adaptive filter of acommunication channel, comprising the steps of: initializing a nominalstep size value and a penalty point value; combining said nominal stepsize value and said penalty point value to generate a step size value;and dynamically changing said step size value in response to acharacteristic measure of a quality of said communication channel,wherein said step size value is changed by adjusting said penalty pointvalue.
 2. The method of claim 1, wherein: said step size value isdecreased by adjusting said penalty point value when a tone originatingfrom the far end of the communication channel is detected.
 3. The methodof claim 1, wherein: said step size value is decreased by adjusting saidpenalty point value when full convergence is achieved.
 4. The method ofclaim 1, wherein: said step size value is decreased by adjusting saidnominal step size value when an achieved combined loss is approximately15 dB or greater.
 5. The method of claim 1, wherein: said step sizevalue is decreased by adjusting said penalty point value when a powerlevel of a residual error signal, P_(e), is less than −60 dBm0 or afar-end channel signal, P_(x), is less than −45 dBm0, that is whenP_(e)<−60 dBm0 or P_(x)<−45 dBm0.
 6. The method of claim 1, wherein:said step size value is decreased by adjusting said penalty point valuewhen a level of said channel's near-end background noise is high.
 7. Themethod of claim 1, wherein: said step size value is decreased byadjusting said penalty point value when weak double-talk in saidcommunication channel is detected.
 8. The method of claim 1, wherein:said step size value is reset by adjusting said nominal step size valuewhen divergence is detected.
 9. The method of claim 1, wherein: saidpenalty point value is reinitialized periodically.
 10. The method ofclaim 9, wherein: said period of reinitializing said penalty point valueis once every 40 samples.
 11. The method of claim 10, wherein: saidperiod corresponds to 5 ms for a 8 kHz sampling rate.
 12. The method ofclaim 10, wherein: said step size value is decreased by adjusting saidpenalty point value when full convergence is achieved.
 13. The method ofclaim 10, wherein: said step size value is increased by adjusting saidnominal step size value when a combined loss exceeds 15 dB.
 14. Themethod of claim 10, wherein: said step size value, μ, is expressed bythe equation log₂(μ)=1+log₂(μ₀)−log₂(f(P_(x)))−log₂(g(N))−ρ, where ρrepresents said penalty point value, μ₀ represents said nominal stepsize value, f(P_(x)) is a function of a far-end power measured withinsaid channel, g(N) is a function of the taps of said adaptive filter;and ρ has a positive or negative integer value of zero, one, or twoassigned to it for every sample within said reinitializing period. 15.The method of claim 7, wherein: said weak double-talk is detected insaid communication channel when a near-end speech signal is at least 6dB less than a far-end speech signal and at least 12 dB above a noisefloor.
 16. The method of claim 12, wherein: said full convergence isachieved when further convergence would be audibly imperceptible tousers of said communication channel.
 17. A method of converging anadaptive filter of a communication channel, comprising the steps of:initializing a nominal step size value and a penalty point value;combining said nominal step size value and said penalty point value togenerate a step size value; and dynamically changing said step sizevalue in response to a characteristic measure of a quality of saidcommunication channel, wherein said step size value is changed byadjusting said nominal step size value.
 18. A method of converging anadaptive filter of a communication channel, according to claim 17,wherein said step size value is selectively changed by adjusting eithersaid nominal step size value, said penalty point value, or both saidnominal step size value and said penalty point value.